[asterisk-users] Sipura phone does not ring

Fran Oliveira tech.oliveira at gmail.com
Wed Dec 6 05:24:14 MST 2006


Sorry for my delay in answer you.
S0 is how pstn line is identified into spa3000. It means that an incoming
call from S0 will be forwarded to "extension at asteriskbox"
yor must configure in sip.conf an account for the sip pstn line and a
context , in extensions.conf the same context with a pattern
or number  for dialing.

example
sip.conf
[100]
host=dynamic
type=friend
context=GW-PSTN-ZEL
secret=xxxxx
qualify=150
authuser=100
username=GW-PSTN-ZEL
accountcode=ZEL-100
port=5061
disallow=all
allow=ulaw

extensions.conf
[GW-PSTN-ZEL]
        exten=>s,1,Answer
        exten=>s,2,NoOp(${CALLERID})
        exten=>s,3,GotoIfTime(10:00-18:00,mon-fri,*,*?HLaboral,s,1)
        exten=>s,4,GotoIfTime(09:00-10:00,mon-fri,*,*?DesvioFax,s,1)
        exten=>s,5,Goto(Cerrado,s,1)

I Have configured the spa 3000 with S0<:s at Asterisk_ip_address>
when spa 3000 receive a call, is forwarded to s extension in asterisk, but
before, it is very important that sip user line be registered with asterisk
for making calls.

I hope  it will help you


2006/11/29, Larry Alkoff <labradley at mindspring.com>:
>
> Fran when you say "specify the next hop" do you mean the S0 line be an
> extension in sip.conf or a context in extensions.conf?
>
> Or should the line simply be tacked on to my [default] context?
>
> Larry
>
> Fran Oliveira wrote:
> > I think it is wrong. You should specify the next hop with some like this
> > S0<:66610 at Asterisk_ip_address>
> >
> >
> >
> > 2006/11/23, Larry Alkoff <labradley at mindspring.com>:
> >>
> >> Problem: SPA3000 phone does not ring for incoming PSTN call although I
> >> can dial out.
> >>
> >> I set up my Sipura with the Voxilla Wizard which is pretty good but
> >> leaves out some important details.
> >>
> >> The Voxilla Wizard for Supura SPA3000 gave me a setting for PSTN Tab ->
> >> Dial Plans ->
> >> Dial Plan 8 (<S0:66610>)
> >>
> >> Should I put extension [66610] in sip.conf with a context in
> >> extensions.conf that will contain dialing instructions?
> >>
> >> Can someone please tell me what the entries under [66610] and the
> >> associated context would look like?
> >>
> >> Or just tell me how to handle this - I'm been stuck for some time with
> >> this.
> >>
> >> The Wizard was nice enough to give detailed settings for sip.conf and
> >> extensions.conf but nothing about to handle Dial Plan 8 except "You'll
> >> need to enter the extension you wish to forward all incoming PSTN calls
> >> to on your Asterisk server". I don't understand how to do that.
> >>
> >> Larry
> >>
> >> --
> >> Larry Alkoff N2LA - Austin TX
>
>
> --
> Larry Alkoff N2LA - Austin TX
> Using Thunderbird on Linux
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061206/c8b954b9/attachment.htm


More information about the asterisk-users mailing list